I did a quick test with this setup. I had a scratch track to record the drums against that was one one track in Reaper. Then I had the midi track to record the drums next to it (with an instance of EZ drummer 2 that's triggered by the drum kit)

Now, the drummer really liked the sounds we were getting out of EZ drummer (better fit to the song), so we tried to record with monitoring on and switching the sounds from the Roland drum brain off.

The problem was latency. My drummer said that on everything but the shortest latency setting, he could really feel the delay. I trust him on that. The problem is that on the shortest latency setting (don't ask me for the buffer size, the Tascam driver offers five settings from "shortest" "to longest") the playback started breaking up and crackling, to a point where it didn't make sense anymore. Now, I figure, I have a bunch of options.

--== lengthy explanation over ==--

1.) Shall we bite the bullet and use the (not so nice) drum sounds from the roland brain (no latency) whilst paying back the scratch track and recording the midi. Would this mean that the recorded midi would be slightly out of time by the amount of the higher latency to guarantee playback?

2.) Is there a drum sampler that uses significantly fewer resources than EZ drummer 2? I was struggling to find any performance data for drum samplers.

3.) Is it worthwhile investing in a different soundcard? Would a cheap Firewire card, e.g. a Maudio FW410 perform better on the same laptop?

4.) (not so preferred option) Would the only way out be a faster computer?

Basically it boils down to windows not being a realtime operating system with no real means provided to users for guaranteed low latency audio performance. It is left up to us to figure out what needs to be done within the narrow confines provided to us since micrsoft long ago took away the option of manually setting up irq's. So then the only practical option left on a windows machine is to disable the most offending hardware devices and hardware features where practical in an effort to give windows less to deal with so that it can better deal with what we care about most, being audio performance. On my laptop for example, I had to disable all cpu powersaving features in the bios, acpi battery method in device manager, and I have to disable wifi only when using low latency audio applications. This isn't a problem with the daw software and plugins. It is a system issue primarily, with other aspects being an audio interface driver issue. If the operating system and an audio interface's driver isn't developed for achieving best low latency performance (and stable performance), there is nothing that can be done further up the chain to the applications and plugins. But with some effort in figuring out what to disable on your machine, you probably can achieve lower latency, even with a not so well-developed driver. And an audio interface with a solid driver definitely helps, too. Also, other than what is mentioned in the article above, if you have a both intel video and another video card, disabling the other video card can be helpful.

Maybe someone else will come along with more to add. But just one more thing: Every system is going to be a crapshoot with windows, because we as users aren't provided adequate control for achieving best audio performance. So then, throwing a new system at it isn't a proper solution and it may or may not work. Either way, it's best to learn how to do what you can with what you got, and you will likely need the same set of skills for making similar changes on a new system. And since laptops are usually more focused around powersaving, you will likely have more trouble with ironing things out on a laptop.

__________________
The media are misleading the public about...pretty much everything.

regarding 1):
that's what I would do to be on the safe side with regard to the performance of the laptop during the recording. Provided that you have well configured your Tascam interface, audio and MIDI data should get recorded in the right location. However, it's always recommended to run round-trip tests (by sending out and re-recording a metronome click track item which had been glued to audio) to ensure that the latency reported to Reaper is correctly take into account. In my setup, I found out after a while that my audio interface didn't report the correct latency, thus, all audio and MIDI items landed slightly off. I ran a few tests and now everything is sample-accurate.

You can always tweak the drum sound after the recording and replace the entire kit with EZDrummer if you like. It may not sound the same during the recording but at least, the recording is flawless.

regarding 3):
how would you connect a Firewire audio interface to your laptop? Does your machine provide a PCMCIA slot to fit in the appropriate adapter card?

Generally, as brainwreck pointed out already, getting near real-time performance on a computer is usually not easy and also not cheap. You can save a lot of money by simply recording with what you have and then editing it afterwards. Real-time capability will probably require a more expensive interface, maybe a dedicated PCIe card inside a desktop computer and a computer with higher specs.

Brainwreck - that article you linked to is really insightful and in depth. Thanks a lot for that. Much better than the usual on the subject.
Now, I have read an article about specifically the changes in Win10 in this regard:https://www.soundonsound.com/reviews...s-10-musicians
And it seems a little less problematic. It touches upon the subject you are reporting too: Wifi or bluetooth interrupting.
But it says that it only works if manufacturers implement this in their drivers. Naturally, my ancient Tascam device doesn't even have Win10 drivers (I think they're from Win8) so naturally it can't support these new functions.
As an aside: it would be interested to know which manufacturers actually have implemented these new functions into their Win10 drivers.

SonicAxiom - Thanks a lot to you too for taking the time. I guess you are right. I should do a round test trip and figure out whether it all lands back in the right place. Rather than chasing an elusive super-low latency system.
(The reason why I'm a bit nervous is that the piece has a lot of tempo changes and as I learned the other day, it can get a bit tricky when midi is recorded under a different time base. I know there's a setting for it in the item properties, but it doesn't play well with takes. That's a subject for another post)

By the way, the old laptop I have has a FW port (no idea what chipset, though) that's why I was curious whether there is anything known like "FW always has lowe latency than USB" or somesuch. There seem to be a lot of preconceptions like that but it's hard to seperate believes from facts.

So, yeah, I guess in summary, unless there is a magic device that has super-well written drivers and doesn't cost the earth, I'll stick with what I have.

Just out of curiosity (and I don't mean to spark a religious debate): Is it true that Macs are generally better at this latency business? I never had one but I remember somebody telling me about their "ring buffer" architecture some time back and why that was superior to the linear buffers in Windows. Is this still the case or have these concepts moved on?

you will probably get better performance as well as higher audio quality by using an RME interface (1st gen Babyface/Fireface). Their driver stability and performance has a very good reputation. You might find a second hand device that suits your needs for a moderate price. This would be a reasonable upgrade (and maybe enables you to even use something like EZDrummer in real-time). You should try to find one to test it on your system. RME interfaces are virtually "unkaputable" and you will get legendary DigiCheck analyzer software on top for free.

I don't know about macs, but linux is an option, too. There has been lots of discussion in the prerelease section of the forum on that. Just say no to windows 10!

Hmm, Linux, I hadn't thought of that. I sit in front of a Linux box in my day job 8-12 hrs a day. I like the system since it gives me the feeling I know what's going on. Having said that, my gut tells me, it's not the best system for the task at hand. Plus, a google search revealed that the Tascam interface I have is not really working under Linux. But if I have some spare time, I'll probably give it a whirl with a live system or so.

Quote:

Originally Posted by SonicAxiom

you will probably get better performance as well as higher audio quality by using an RME interface (1st gen Babyface/Fireface). Their driver stability and performance has a very good reputation. You might find a second hand device that suits your needs for a moderate price. This would be a reasonable upgrade (and maybe enables you to even use something like EZDrummer in real-time). You should try to find one to test it on your system. RME interfaces are virtually "unkaputable" and you will get legendary DigiCheck analyzer software on top for free.

Yeah, the RME interfaces have a good reputation for their support. Even older ones still get driver updates and would make me confident that the latest OS changes are actually implemented in the driver.
(Mind you - Behringer also releases regular updates for their interfaces, I couldn't find any changelogs though)
* ETA: That's not true, I just found Behringers driver changelog.

The problem is that my budget for this little exercise is more towards the Behringer end than the (even second hand) RME end. I guess, if you have an RME, you hang on to it.